• Journal of Communications and Networks
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• v.16 no.3
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• pp.344-354
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• 2014

Transmission control protocol friendly rate control (TFRC) is designed to mainly provide optimal service for unicast applications, such as multimedia streaming in the best-effort Internet environment. However, high bandwidth networks with large delays present an environment where TFRC may have a problem in utilizing the full bandwidth. TFRC inherits the slow-start mechanism of TCP Reno, but this is a time-consuming process that may require many round-trip-times (RTTs), until an appropriate sending rate is reached. Another disadvantage inherited from TCP Reno is the RTT-unfairness problem, which severely affects the performance of long-RTT flows. In this paper, we suggest enhanced TFRC for high quality video streaming over high bandwidth delay product networks. First, we propose a fast startup scheme that increases the data rate more aggressively than the slow-start, while mitigating the overshooting problem. Second, we propose a bandwidth estimation method to achieve more equitable bandwidth allocations among streaming flows that compete for the same narrow link with different RTTs. Finally, we improve the responsiveness of TFRC in the presence of severe congestion. Simulation results have shown that our proposal can achieve a fast startup and provide fairness with competing flows compared to the original TFRC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.9B
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• pp.829-837
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• 2006

Due to the internet network congestion, packets may be dropped or delayed at routers. This phenomenon degrades the quality of streaming applications that require high QoS requirements. The proposed algorithm in this paper, called DBRC(Delay-Based Rate Control), tries to cause router queue occupancy to reach a steady state or equilibrium by throttling the transmission rate of the multimedia traffics when network delays tend to increase and also probing for more bandwidth when network delays tend to decrease. Simulation results show that the proposed algorithm provides smooth transmission rate, nearly constant delay and low packet loss rates, compared with TFRC(TCP Friendly Rate Control) that is one of dominant multimedia congestion control algorithms.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.749-750
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• 2006

TFRC is an equation-based rate control scheme originally developed for video transmission over wired networks. When applied to the wireless networks, it suffers from performance degradation. In this thesis, we propose an end-to-end loss discrimination algorithm to improve the performance of TFRC over wireless networks. The proposed WLD-TFRC scheme combines Spike and WLD(Wireless Loss Discount) algorithms to discriminate wireless loss from congestion loss, and to discount feedback loss event rate. Experimental results show that WLD-TFRC outperforms the original TFRC and effectively reduce the degradation of the video quality caused by the wireless link status.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1609-1617
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• 2013

TFRC(TCP-Friendly Rate Control) has a performance degradation in wireless networks because it performs congestion control by judging all the losses occurred in wireless networks as a congestion indicator. It is also degraded by the increased Round Trip Time(RTT) due to packet retransmission and contention overhead in the link layer. In this paper, we propose an adaptive rate control scheme based on cross-layer to improve the quality of streaming services in the wireless networks. It provides new RTT estimation and loss discrimination methods to improve transmission rate of TFRC. The simulation results show that the proposed scheme can improve the performance of TFRC.

• Journal of KIISE:Information Networking
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• v.29 no.5
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• pp.495-500
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• 2002

We propose a new estimation method for rate adjustment in the face of a packet loss in the TFRC protocol, a TCP-Friendly congestion control protocol for UDP flows. Previous methods respond in a sensitive way to a single packet loss, resulting in oscillatory transmission behavior. This is an undesirable for multimedia services demanding constant bandwidth. The proposed TFRC provides more smooth and fair (against TCP flows) transmission through collective response based on multiple packets loss events. We show our "Exponential smoothing method" performs better than known "Weight smoothing method" in terms of smoothness and fairness.

• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.679-681
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• 2001

인터넷 화상 전화와 같이 비디오 데이터를 압축하여 실시간 전송하는 응용 프로그램의 사용 시 일어날 수 있는 비디오 데이터 오류 전파를 방지하기 위한 대표적인 기법으로 FEC를 향상시킨 Periodic FEC가 있다. 본 논문에서는 ACK 메시지를 기반으로 연속적인 패킷 손실을 예측하여 PFEC의 중복 정보(redundant information)외 전송량 조절이 가능한ACK기반 중복 정보 제어 기법을 제안한다. 또한 대용량의 멀티미디어 데이터 전송 시 발생할 수 있는 congestion을 제어하기 위해 TCP와 공정하게 대역폭을 공유하는 TCP-friendly rate control 기법을 고려하여 PFEC의 중복 정보의 양을 네트워크 상태에 적응적으로 조절하는 기법을 제안한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2B
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• pp.44-52
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• 2005

This paper proposes a neural network prediction model to improve the valid packet transmission rate for the QoS(Quality of Service) of multimedia transmission. The Round Trip Time(RTT) and Packet Loss Rate(PLR) are predicted using a neural network and then the transmission rate is decided based on the predicted RTT and the PLR. The suggested method will improve the transmission rate since it uses the rate control factors corresponding to time of data is being transmitted, while the conventional one uses the transmission rate determined based on the past informations. An experimental set-up has been established using a Linux PC system, and the multimedia data are transmitted using UDP protocol in real time. The valid transmitted packets are about 5% higher than the one in the conventional TCP-Friendly congestion control method when the suggested algorithm was applied.

• Proceedings of the Korean Information Science Society Conference
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• 2000.04a
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• pp.271-273
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• 2000

본 논문에서는 TCP 수신자가 지연 ACK 옵션(Delayed ACK Option)을 사용할 경우에 TCP 송신자에게 발생하는 성능 저하요인들을 분석하고 다음과 같은 해결책을 제시하였다. 먼저, 느린 시작 구간(Slow Start phase) 처음에 생기는 ACK 타임아웃은 큰 초기 윈도우(large initial window)또는 1-bit 마킹 기법을 통해 해결할 수 있다. 그리고, 느린 시작 구간과 혼잡 회피 구간(Congestion Avoidance phase)에서 혼잡 윈도우(cwnd)가 천천히 증가하는 문제는 적절히 바이트 카운팅 기법을 사용함으로써 해결할 수 있다. 마지막으로, 송신자가 버스트(burst)한 데이터를 네트웍에 발생시키는 문제는 트래픽을 평활(pacing)함으로써 해결할 수 있다. 또한 본 연구에서는 분석적 모델링을 통하여 TCP가 보내는 평균 전송률을 구하였으며 이 결과는 TCP에 친화한 전송률 기반 전송방법(TCP Friendly Rate Based Control)에 응용될 수 있을 것이다. 그리고 시뮬레이션을 통해서 제시한 방법의 성능이 향상됨을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5B
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• pp.517-525
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• 2009

We propose a quality guaranteed scalable video streaming service over the Internet using a new rate adaptation algorithm. Because video data requires much more bandwidth rather than other types of service, therefore, quality of video streaming service should be guaranteed while providing friendliness with other service flows over the Internet. To successfully provide this, we propose a framework for providing quality-guaranteed streaming service using two-channel transport layer and rate adaptation of scalable video stream. In this framework, baseline layer for scalable video is transmitted using TCP transport for minimum qualify service. Enhancement layers are delivered using TFRC transport with layer adaptation algorithm. The proposed framework jointly uses the status of playout buffer in the client and the encoding rate of layers in media contents. Therefore, the proposed algorithm can remarkably guarantee minimum quality of streaming service rather than conventional approaches regardless of network congestion and the encoding rate variation of media content.

• Journal of Advanced Navigation Technology
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• v.14 no.3
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• pp.358-366
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• 2010

In the wireless network the congestion and delay occurs mainly when there are too many packets for the network to process or the sender transmits more packets than the receiver can accept. The congestion and delay is the reason of packet loss which degrades the performance of multimedia streaming. This paper proposes a novel transmission rate monitoring-based optimization mechanism to optimize packet loss and to improve QoS. The proposed scheme is based on the trade-off relationship between transmission rate monitoring and overhead monitoring. For this purpose this paper processes a source rate control-based optimization which optimizes congestion and delay. Performance evaluated RED, TFRC, and the proposed mechanism. The simulation results show that the proposed mechanism is more efficient than REC(Random Early Detection) mechanism and TFRC(TCP-friendly Rate Control) mechanism in packet loss rate, throughput rate, and average response rate.